Connectivity and voice quality are key reliability issues in today's VoIP (voice over Internet Protocol) networks. Today's VoIP users increasingly expect the Quality of Service (QoS) of the call to be equal or close to that of the Public Switched Telephone Network (PSTN). Because network conditions may change rapidly and continuously, connectivity for a VoIP call cannot be guaranteed. Further, IP network problems such as packet loss, packet delay, and out of order delivery may lead to deteriorating quality of VoIP voice calls.
Unlike data connections, a real-time application like voice calls place much stricter requirements on packet delivery sequence and time. Significant packet loss, packet delay, and out of order delivery problems make telephone conversations difficult. Users may experience echoes and talk overlap that are perceived as significant indicators of inferior QoS.
Previously, voice call routing only exists in the voice network domain, which is characterized by logical trunk groups interconnecting switches in the network. Voice routing in this context involves routing on circuits or logical trunk groups. The IP network routing function previously only operate in the data network domain and is not operable with the applications operating in the application layer. Therefore, routing voice calls on a data network to satisfy QoS remained a challenge unconquered by the industry.